platform-external-webrtc

Files

ossu c3d4b48e7e Store/restore RTP state for audio streams with same SSRC within a call

This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.

BUG=webrtc:7631

Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}

2017-05-23 13:07:11 +00:00

api

Renaming probing_interval to bwe_period globally.

2017-05-18 21:32:41 +00:00

audio

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

base

Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.

2017-05-22 22:32:51 +00:00

call

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

examples

AppRTCMobile support to turn off the WebRTC AGC and HPF

2017-05-18 22:09:33 +00:00

logging

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

media

Field trial support to whenever possible turn off the AGC and HPF

2017-05-22 22:48:47 +00:00

modules

Preserve level controller output when no other effects are active

2017-05-23 12:33:56 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

2017-05-19 20:09:47 +00:00

sdk

Field trial support to whenever possible turn off the AGC and HPF

2017-05-22 22:48:47 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Adding target to track asm_defines.h

2017-05-15 21:59:17 +00:00

test

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

tools

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

video

Reland of use allocated encoders in SimulcastEncoderAdapter. (patchset #1 id:1 of https://codereview.webrtc.org/2893003002/ )

2017-05-23 10:32:16 +00:00

voice_engine

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

.gitignore

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BUILD.gn

Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )

2017-05-17 13:52:58 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

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common_types.cc

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +00:00

common_types.h

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_frame.h

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

video_send_stream.h

Update adaptation stats to support degradations in both resolution and framerate.

2017-05-16 06:40:18 +00:00

webrtc.gni

Replace sanitizers:deps with exe_and_shlib_deps (webrtc)

2017-05-19 06:51:46 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.